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1 # Copyright 2014 The Chromium Authors. All rights reserved. | 1 # Copyright 2014 The Chromium Authors. All rights reserved. |
2 # Use of this source code is governed by a BSD-style license that can be | 2 # Use of this source code is governed by a BSD-style license that can be |
3 # found in the LICENSE file. | 3 # found in the LICENSE file. |
4 | 4 |
5 import("//build/config/features.gni") | 5 import("//build/config/features.gni") |
6 | 6 |
7 # From third_party/libjingle/libjingle.gyp's target_defaults. | 7 # From third_party/libjingle/libjingle.gyp's target_defaults. |
8 config("jingle_unexported_configs") { | 8 config("jingle_unexported_configs") { |
9 defines = [ | |
10 "EXPAT_RELATIVE_PATH", | |
11 "FEATURE_ENABLE_SSL", | |
12 "GTEST_RELATIVE_PATH", | |
13 "HAVE_OPENSSL_SSL_H", | |
14 "HAVE_SCTP", | |
15 "HAVE_SRTP", | |
16 "HAVE_WEBRTC_VIDEO", | |
17 "HAVE_WEBRTC_VOICE", | |
18 "LOGGING_INSIDE_WEBRTC", | |
19 "NO_MAIN_THREAD_WRAPPING", | |
20 "NO_SOUND_SYSTEM", | |
21 "SRTP_RELATIVE_PATH", | |
22 "SSL_USE_OPENSSL", | |
23 "USE_WEBRTC_DEV_BRANCH", | |
24 "ENABLE_EXTERNAL_AUTH", | |
25 "WEBRTC_CHROMIUM_BUILD", | |
26 ] | |
27 | |
28 include_dirs = [ | 9 include_dirs = [ |
29 "overrides", | 10 "overrides", |
30 "../../third_party/webrtc_overrides", | 11 "../../third_party/webrtc_overrides", |
31 "source", | 12 "source", |
32 "../../testing/gtest/include", | 13 "../../testing/gtest/include", |
33 "../../third_party", | 14 "../../third_party", |
34 "../../third_party/libyuv/include", | 15 "../../third_party/libyuv/include", |
35 "../../third_party/usrsctp/usrsctplib", | 16 "../../third_party/usrsctp/usrsctplib", |
36 ] | 17 ] |
37 } | 18 } |
38 | 19 |
39 # From third_party/libjingle/libjingle.gyp's target_defaults. | 20 # From third_party/libjingle/libjingle.gyp's target_defaults. |
40 config("jingle_public_configs") { | 21 config("jingle_public_configs") { |
41 include_dirs = [ | 22 include_dirs = [ |
42 "../../third_party/webrtc_overrides", | 23 "../../third_party/webrtc_overrides", |
43 "overrides", | 24 "overrides", |
44 "source", | 25 "source", |
45 "../../testing/gtest/include", | 26 "../../testing/gtest/include", |
46 "../../third_party", | 27 "../../third_party", |
47 ] | 28 ] |
48 defines = [ | |
49 "FEATURE_ENABLE_SSL", | |
50 "FEATURE_ENABLE_VOICEMAIL", | |
51 "EXPAT_RELATIVE_PATH", | |
52 "GTEST_RELATIVE_PATH", | |
53 "NO_MAIN_THREAD_WRAPPING", | |
54 "NO_SOUND_SYSTEM", | |
55 ] | |
56 | |
57 if (is_linux) { | |
58 defines += [ | |
59 "LINUX", | |
60 "WEBRTC_LINUX", | |
61 ] | |
62 } | |
63 if (is_mac) { | |
64 defines += [ | |
65 "OSX", | |
66 "WEBRTC_MAC", | |
67 ] | |
68 } | |
69 if (is_ios) { | |
70 defines += [ | |
71 "IOS", | |
72 "WEBRTC_MAC", | |
73 "WEBRTC_IOS", | |
74 ] | |
75 } | |
76 if (is_win) { | |
77 defines += [ "WEBRTC_WIN" ] | |
78 } | |
79 if (is_android) { | |
80 defines += [ "ANDROID" ] | |
81 } | |
82 if (is_posix) { | |
83 defines += [ "WEBRTC_POSIX" ] | |
84 } | |
85 if (is_chromeos) { | |
86 defines += [ "CHROMEOS" ] | |
87 } | |
88 } | 29 } |
89 | 30 |
90 # From third_party/libjingle/libjingle.gyp's target_defaults. | 31 # From third_party/libjingle/libjingle.gyp's target_defaults. |
91 group("jingle_deps") { | 32 group("jingle_deps") { |
92 public_deps = [ | 33 public_deps = [ |
93 "//third_party/expat", | 34 "//third_party/expat", |
94 ] | 35 ] |
95 deps = [ | 36 deps = [ |
96 "//base", | 37 "//base", |
97 "//crypto:platform", | 38 "//crypto:platform", |
98 "//net", | 39 "//net", |
99 ] | 40 ] |
100 } | 41 } |
101 | 42 |
102 # GYP version: third_party/libjingle.gyp:libjingle | 43 # GYP version: third_party/libjingle.gyp:libjingle |
103 static_library("libjingle") { | 44 static_library("libjingle") { |
104 p2p_dir = "../webrtc/p2p" | |
105 xmllite_dir = "../webrtc/libjingle/xmllite" | |
106 xmpp_dir = "../webrtc/libjingle/xmpp" | |
107 sources = [ | |
108 # List from third_party/libjingle/libjingle_common.gypi | |
109 "$p2p_dir/base/asyncstuntcpsocket.cc", | |
110 "$p2p_dir/base/asyncstuntcpsocket.h", | |
111 "$p2p_dir/base/basicpacketsocketfactory.cc", | |
112 "$p2p_dir/base/basicpacketsocketfactory.h", | |
113 "$p2p_dir/base/candidate.h", | |
114 "$p2p_dir/base/common.h", | |
115 "$p2p_dir/base/dtlstransport.h", | |
116 "$p2p_dir/base/dtlstransportchannel.cc", | |
117 "$p2p_dir/base/dtlstransportchannel.h", | |
118 "$p2p_dir/base/p2pconstants.cc", | |
119 "$p2p_dir/base/p2pconstants.h", | |
120 "$p2p_dir/base/p2ptransport.cc", | |
121 "$p2p_dir/base/p2ptransport.h", | |
122 "$p2p_dir/base/p2ptransportchannel.cc", | |
123 "$p2p_dir/base/p2ptransportchannel.h", | |
124 "$p2p_dir/base/port.cc", | |
125 "$p2p_dir/base/port.h", | |
126 "$p2p_dir/base/portallocator.cc", | |
127 "$p2p_dir/base/portallocator.h", | |
128 "$p2p_dir/base/pseudotcp.cc", | |
129 "$p2p_dir/base/pseudotcp.h", | |
130 "$p2p_dir/base/rawtransport.cc", | |
131 "$p2p_dir/base/rawtransport.h", | |
132 "$p2p_dir/base/rawtransportchannel.cc", | |
133 "$p2p_dir/base/rawtransportchannel.h", | |
134 "$p2p_dir/base/relayport.cc", | |
135 "$p2p_dir/base/relayport.h", | |
136 "$p2p_dir/base/session.cc", | |
137 "$p2p_dir/base/session.h", | |
138 "$p2p_dir/base/sessiondescription.cc", | |
139 "$p2p_dir/base/sessiondescription.h", | |
140 "$p2p_dir/base/sessionid.h", | |
141 "$p2p_dir/base/stun.cc", | |
142 "$p2p_dir/base/stun.h", | |
143 "$p2p_dir/base/stunport.cc", | |
144 "$p2p_dir/base/stunport.h", | |
145 "$p2p_dir/base/stunrequest.cc", | |
146 "$p2p_dir/base/stunrequest.h", | |
147 "$p2p_dir/base/tcpport.cc", | |
148 "$p2p_dir/base/tcpport.h", | |
149 "$p2p_dir/base/transport.cc", | |
150 "$p2p_dir/base/transport.h", | |
151 "$p2p_dir/base/transportchannel.cc", | |
152 "$p2p_dir/base/transportchannel.h", | |
153 "$p2p_dir/base/transportchannelimpl.h", | |
154 "$p2p_dir/base/transportcontroller.cc", | |
155 "$p2p_dir/base/transportcontroller.h", | |
156 "$p2p_dir/base/transportdescription.cc", | |
157 "$p2p_dir/base/transportdescription.h", | |
158 "$p2p_dir/base/transportdescriptionfactory.cc", | |
159 "$p2p_dir/base/transportdescriptionfactory.h", | |
160 "$p2p_dir/base/turnport.cc", | |
161 "$p2p_dir/base/turnport.h", | |
162 "$p2p_dir/client/basicportallocator.cc", | |
163 "$p2p_dir/client/basicportallocator.h", | |
164 "$p2p_dir/client/httpportallocator.cc", | |
165 "$p2p_dir/client/httpportallocator.h", | |
166 "$p2p_dir/client/socketmonitor.cc", | |
167 "$p2p_dir/client/socketmonitor.h", | |
168 "$xmllite_dir/qname.cc", | |
169 "$xmllite_dir/qname.h", | |
170 "$xmllite_dir/xmlbuilder.cc", | |
171 "$xmllite_dir/xmlbuilder.h", | |
172 "$xmllite_dir/xmlconstants.cc", | |
173 "$xmllite_dir/xmlconstants.h", | |
174 "$xmllite_dir/xmlelement.cc", | |
175 "$xmllite_dir/xmlelement.h", | |
176 "$xmllite_dir/xmlnsstack.cc", | |
177 "$xmllite_dir/xmlnsstack.h", | |
178 "$xmllite_dir/xmlparser.cc", | |
179 "$xmllite_dir/xmlparser.h", | |
180 "$xmllite_dir/xmlprinter.cc", | |
181 "$xmllite_dir/xmlprinter.h", | |
182 "$xmpp_dir/asyncsocket.h", | |
183 "$xmpp_dir/constants.cc", | |
184 "$xmpp_dir/constants.h", | |
185 "$xmpp_dir/jid.cc", | |
186 "$xmpp_dir/jid.h", | |
187 "$xmpp_dir/plainsaslhandler.h", | |
188 "$xmpp_dir/prexmppauth.h", | |
189 "$xmpp_dir/saslcookiemechanism.h", | |
190 "$xmpp_dir/saslhandler.h", | |
191 "$xmpp_dir/saslmechanism.cc", | |
192 "$xmpp_dir/saslmechanism.h", | |
193 "$xmpp_dir/saslplainmechanism.h", | |
194 "$xmpp_dir/xmppclient.cc", | |
195 "$xmpp_dir/xmppclient.h", | |
196 "$xmpp_dir/xmppclientsettings.h", | |
197 "$xmpp_dir/xmppengine.h", | |
198 "$xmpp_dir/xmppengineimpl.cc", | |
199 "$xmpp_dir/xmppengineimpl.h", | |
200 "$xmpp_dir/xmppengineimpl_iq.cc", | |
201 "$xmpp_dir/xmpplogintask.cc", | |
202 "$xmpp_dir/xmpplogintask.h", | |
203 "$xmpp_dir/xmppstanzaparser.cc", | |
204 "$xmpp_dir/xmppstanzaparser.h", | |
205 "$xmpp_dir/xmpptask.cc", | |
206 "$xmpp_dir/xmpptask.h", | |
207 ] | |
208 | |
209 # TODO(jschuh): crbug.com/167187 fix size_t to int truncations. | 45 # TODO(jschuh): crbug.com/167187 fix size_t to int truncations. |
210 configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] | 46 configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] |
211 | 47 |
212 public_deps = [ | 48 public_deps = [ |
213 ":jingle_deps", | 49 ":jingle_deps", |
214 ] | 50 ] |
215 deps = [ | 51 deps = [ |
216 "//third_party/webrtc/base:rtc_base", | 52 "//third_party/webrtc/base:rtc_base", |
53 "//third_party/webrtc/libjingle/xmllite", | |
tommi (sloooow) - chröme
2016/05/31 13:36:09
out of curiosity - do you know what projects depen
kjellander_chromium
2016/05/31 21:20:19
I believe it's only src/jingle and src/remoting. I
| |
54 "//third_party/webrtc/libjingle/xmpp", | |
55 "//third_party/webrtc/p2p:rtc_p2p", | |
217 ] | 56 ] |
218 | 57 |
219 # From libjingle_common.gypi's conditions list. | 58 # From libjingle_common.gypi's conditions list. |
220 if (is_win) { | 59 if (is_win) { |
221 cflags = [ "/wd4005" ] | 60 cflags = [ "/wd4005" ] |
222 } | 61 } |
223 | 62 |
224 if (is_nacl) { | 63 if (is_nacl) { |
225 # For NACL, we have to add a default implementation for field_trail. | 64 # For NACL, we have to add a default implementation for field_trail. |
226 deps += [ | 65 deps += [ |
227 "//native_client_sdk/src/libraries/nacl_io", | 66 "//native_client_sdk/src/libraries/nacl_io", |
228 "//third_party/webrtc/system_wrappers:field_trial_default", | 67 "//third_party/webrtc/system_wrappers:field_trial_default", |
229 ] | 68 ] |
230 } else { | 69 } else { |
231 # Otherwise, we just add the field_trial which redirects to base. | 70 # Otherwise, we just add the field_trial which redirects to base. |
232 sources += [ "overrides/field_trial.cc" ] | 71 sources = [ |
72 "overrides/field_trial.cc", | |
73 ] | |
233 } | 74 } |
234 | 75 |
235 configs += [ ":jingle_unexported_configs" ] | 76 configs += [ |
236 public_configs = [ ":jingle_public_configs" ] | 77 ":jingle_unexported_configs", |
78 "//third_party/webrtc:common_config", | |
79 ] | |
80 public_configs = [ | |
81 ":jingle_public_configs", | |
82 "//third_party/webrtc:common_inherited_config", | |
83 ] | |
237 } | 84 } |
238 | 85 |
239 if (enable_webrtc) { | 86 if (enable_webrtc) { |
240 source_set("libjingle_webrtc") { | 87 source_set("libjingle_webrtc") { |
241 sources = [ | 88 sources = [ |
242 "overrides/init_webrtc.cc", | 89 "overrides/init_webrtc.cc", |
243 "overrides/init_webrtc.h", | 90 "overrides/init_webrtc.h", |
244 ] | 91 ] |
245 configs += [ ":jingle_unexported_configs" ] | 92 configs += [ |
246 public_configs = [ ":jingle_public_configs" ] | 93 ":jingle_unexported_configs", |
94 "//third_party/webrtc:common_config", | |
95 ] | |
96 public_configs = [ | |
97 ":jingle_public_configs", | |
98 "//third_party/webrtc:common_inherited_config", | |
99 ] | |
247 public_deps = [ | 100 public_deps = [ |
248 ":libjingle_webrtc_common", | 101 ":libjingle_webrtc_common", |
249 ] | 102 ] |
250 } | 103 } |
251 | 104 |
252 source_set("libjingle_webrtc_common") { | 105 source_set("libjingle_webrtc_common") { |
253 sources = [ | |
254 "../webrtc/api/audiotrack.cc", | |
255 "../webrtc/api/audiotrack.h", | |
256 "../webrtc/api/datachannel.cc", | |
257 "../webrtc/api/datachannel.h", | |
258 "../webrtc/api/dtlsidentitystore.cc", | |
259 "../webrtc/api/dtlsidentitystore.h", | |
260 "../webrtc/api/dtmfsender.cc", | |
261 "../webrtc/api/dtmfsender.h", | |
262 "../webrtc/api/jsep.h", | |
263 "../webrtc/api/jsepicecandidate.cc", | |
264 "../webrtc/api/jsepicecandidate.h", | |
265 "../webrtc/api/jsepsessiondescription.cc", | |
266 "../webrtc/api/jsepsessiondescription.h", | |
267 "../webrtc/api/localaudiosource.cc", | |
268 "../webrtc/api/localaudiosource.h", | |
269 "../webrtc/api/mediaconstraintsinterface.cc", | |
270 "../webrtc/api/mediaconstraintsinterface.h", | |
271 "../webrtc/api/mediacontroller.cc", | |
272 "../webrtc/api/mediacontroller.h", | |
273 "../webrtc/api/mediastream.cc", | |
274 "../webrtc/api/mediastream.h", | |
275 "../webrtc/api/mediastreamhandler.cc", | |
276 "../webrtc/api/mediastreamhandler.h", | |
277 "../webrtc/api/mediastreaminterface.h", | |
278 "../webrtc/api/mediastreamobserver.cc", | |
279 "../webrtc/api/mediastreamobserver.h", | |
280 "../webrtc/api/mediastreamprovider.h", | |
281 "../webrtc/api/mediastreamproxy.h", | |
282 "../webrtc/api/mediastreamtrack.h", | |
283 "../webrtc/api/mediastreamtrackproxy.h", | |
284 "../webrtc/api/notifier.h", | |
285 "../webrtc/api/peerconnection.cc", | |
286 "../webrtc/api/peerconnection.h", | |
287 "../webrtc/api/peerconnectionfactory.cc", | |
288 "../webrtc/api/peerconnectionfactory.h", | |
289 "../webrtc/api/peerconnectioninterface.h", | |
290 "../webrtc/api/portallocatorfactory.cc", | |
291 "../webrtc/api/portallocatorfactory.h", | |
292 "../webrtc/api/remoteaudiosource.cc", | |
293 "../webrtc/api/remoteaudiosource.h", | |
294 "../webrtc/api/remoteaudiotrack.cc", | |
295 "../webrtc/api/remoteaudiotrack.h", | |
296 "../webrtc/api/rtpreceiver.cc", | |
297 "../webrtc/api/rtpreceiver.h", | |
298 "../webrtc/api/rtpreceiverinterface.h", | |
299 "../webrtc/api/rtpsender.cc", | |
300 "../webrtc/api/rtpsender.h", | |
301 "../webrtc/api/rtpsenderinterface.h", | |
302 "../webrtc/api/sctputils.cc", | |
303 "../webrtc/api/sctputils.h", | |
304 "../webrtc/api/statscollector.cc", | |
305 "../webrtc/api/statscollector.h", | |
306 "../webrtc/api/statstypes.cc", | |
307 "../webrtc/api/statstypes.h", | |
308 "../webrtc/api/streamcollection.h", | |
309 "../webrtc/api/umametrics.h", | |
310 "../webrtc/api/videocapturertracksource.cc", | |
311 "../webrtc/api/videocapturertracksource.h", | |
312 "../webrtc/api/videosourceproxy.h", | |
313 "../webrtc/api/videotrack.cc", | |
314 "../webrtc/api/videotrack.h", | |
315 "../webrtc/api/videotracksource.cc", | |
316 "../webrtc/api/videotracksource.h", | |
317 "../webrtc/api/webrtcsdp.cc", | |
318 "../webrtc/api/webrtcsdp.h", | |
319 "../webrtc/api/webrtcsession.cc", | |
320 "../webrtc/api/webrtcsession.h", | |
321 "../webrtc/api/webrtcsessiondescriptionfactory.cc", | |
322 "../webrtc/api/webrtcsessiondescriptionfactory.h", | |
323 "../webrtc/media/base/audiorenderer.h", | |
324 "../webrtc/media/base/codec.cc", | |
325 "../webrtc/media/base/codec.h", | |
326 "../webrtc/media/base/cryptoparams.h", | |
327 "../webrtc/media/base/hybriddataengine.h", | |
328 "../webrtc/media/base/mediachannel.h", | |
329 "../webrtc/media/base/mediaconstants.cc", | |
330 "../webrtc/media/base/mediaconstants.h", | |
331 "../webrtc/media/base/mediaengine.cc", | |
332 "../webrtc/media/base/mediaengine.h", | |
333 "../webrtc/media/base/rtpdataengine.cc", | |
334 "../webrtc/media/base/rtpdataengine.h", | |
335 "../webrtc/media/base/rtpdump.cc", | |
336 "../webrtc/media/base/rtpdump.h", | |
337 "../webrtc/media/base/rtputils.cc", | |
338 "../webrtc/media/base/rtputils.h", | |
339 "../webrtc/media/base/streamparams.cc", | |
340 "../webrtc/media/base/streamparams.h", | |
341 "../webrtc/media/base/turnutils.cc", | |
342 "../webrtc/media/base/turnutils.h", | |
343 "../webrtc/media/base/videoadapter.cc", | |
344 "../webrtc/media/base/videoadapter.h", | |
345 "../webrtc/media/base/videobroadcaster.cc", | |
346 "../webrtc/media/base/videobroadcaster.h", | |
347 "../webrtc/media/base/videocapturer.cc", | |
348 "../webrtc/media/base/videocapturer.h", | |
349 "../webrtc/media/base/videocommon.cc", | |
350 "../webrtc/media/base/videocommon.h", | |
351 "../webrtc/media/base/videoframe.cc", | |
352 "../webrtc/media/base/videoframe.h", | |
353 "../webrtc/media/base/videoframefactory.cc", | |
354 "../webrtc/media/base/videoframefactory.h", | |
355 "../webrtc/media/base/videosourcebase.cc", | |
356 "../webrtc/media/base/videosourcebase.h", | |
357 "../webrtc/media/engine/simulcast.cc", | |
358 "../webrtc/media/engine/simulcast.h", | |
359 "../webrtc/media/engine/webrtccommon.h", | |
360 "../webrtc/media/engine/webrtcmediaengine.cc", | |
361 "../webrtc/media/engine/webrtcmediaengine.h", | |
362 "../webrtc/media/engine/webrtcvideoengine2.cc", | |
363 "../webrtc/media/engine/webrtcvideoengine2.h", | |
364 "../webrtc/media/engine/webrtcvideoframe.cc", | |
365 "../webrtc/media/engine/webrtcvideoframe.h", | |
366 "../webrtc/media/engine/webrtcvideoframefactory.cc", | |
367 "../webrtc/media/engine/webrtcvideoframefactory.h", | |
368 "../webrtc/media/engine/webrtcvoe.h", | |
369 "../webrtc/media/engine/webrtcvoiceengine.cc", | |
370 "../webrtc/media/engine/webrtcvoiceengine.h", | |
371 "../webrtc/media/sctp/sctpdataengine.cc", | |
372 "../webrtc/media/sctp/sctpdataengine.h", | |
373 "../webrtc/pc/audiomonitor.cc", | |
374 "../webrtc/pc/audiomonitor.h", | |
375 "../webrtc/pc/bundlefilter.cc", | |
376 "../webrtc/pc/bundlefilter.h", | |
377 "../webrtc/pc/channel.cc", | |
378 "../webrtc/pc/channel.h", | |
379 "../webrtc/pc/channelmanager.cc", | |
380 "../webrtc/pc/channelmanager.h", | |
381 "../webrtc/pc/currentspeakermonitor.cc", | |
382 "../webrtc/pc/currentspeakermonitor.h", | |
383 "../webrtc/pc/externalhmac.cc", | |
384 "../webrtc/pc/externalhmac.h", | |
385 "../webrtc/pc/mediamonitor.cc", | |
386 "../webrtc/pc/mediamonitor.h", | |
387 "../webrtc/pc/mediasession.cc", | |
388 "../webrtc/pc/mediasession.h", | |
389 "../webrtc/pc/mediasink.h", | |
390 "../webrtc/pc/rtcpmuxfilter.cc", | |
391 "../webrtc/pc/rtcpmuxfilter.h", | |
392 "../webrtc/pc/srtpfilter.cc", | |
393 "../webrtc/pc/srtpfilter.h", | |
394 "../webrtc/pc/voicechannel.h", | |
395 ] | |
396 | |
397 configs -= [ "//build/config/compiler:chromium_code" ] | 106 configs -= [ "//build/config/compiler:chromium_code" ] |
398 configs += [ "//build/config/compiler:no_chromium_code" ] | 107 configs += [ "//build/config/compiler:no_chromium_code" ] |
399 | 108 |
400 configs += [ ":jingle_unexported_configs" ] | 109 configs += [ |
401 public_configs = [ ":jingle_public_configs" ] | 110 ":jingle_unexported_configs", |
111 "//third_party/webrtc:common_config", | |
112 ] | |
113 public_configs = [ | |
114 ":jingle_public_configs", | |
115 "//third_party/webrtc:common_inherited_config", | |
116 ] | |
402 | 117 |
403 deps = [ | 118 deps = [ |
404 ":libjingle", | 119 ":libjingle", |
405 "//third_party/libsrtp", | 120 "//third_party/libsrtp", |
406 "//third_party/usrsctp", | 121 "//third_party/usrsctp", |
407 "//third_party/webrtc", | 122 "//third_party/webrtc", |
123 "//third_party/webrtc/api:libjingle_peerconnection", | |
124 "//third_party/webrtc/media:rtc_media", | |
408 "//third_party/webrtc/modules/media_file", | 125 "//third_party/webrtc/modules/media_file", |
409 "//third_party/webrtc/modules/video_capture", | 126 "//third_party/webrtc/modules/video_capture", |
127 "//third_party/webrtc/pc:rtc_pc", | |
410 "//third_party/webrtc/system_wrappers", | 128 "//third_party/webrtc/system_wrappers", |
411 "//third_party/webrtc/voice_engine", | 129 "//third_party/webrtc/voice_engine", |
412 ] | 130 ] |
413 } | 131 } |
414 | |
415 source_set("libstunprober") { | |
416 p2p_dir = "../webrtc/p2p" | |
417 sources = [ | |
418 "$p2p_dir/stunprober/stunprober.cc", | |
419 ] | |
420 | |
421 deps = [ | |
422 ":libjingle_webrtc_common", | |
423 "//third_party/webrtc/base:rtc_base", | |
424 ] | |
425 } | |
426 } # enable_webrtc | 132 } # enable_webrtc |
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